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af_afade.c
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1 /*
2  * Copyright (c) 2013-2015 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 /**
22  * @file
23  * fade audio filter
24  */
25 
26 #include "libavutil/audio_fifo.h"
27 #include "libavutil/opt.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "internal.h"
31 
32 typedef struct {
33  const AVClass *class;
34  int type;
35  int curve, curve2;
36  int nb_samples;
37  int64_t start_sample;
38  int64_t duration;
39  int64_t start_time;
40  int overlap;
41  int cf0_eof;
42  int crossfade_is_over;
43  AVAudioFifo *fifo[2];
44  int64_t pts;
45 
46  void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
47  int nb_samples, int channels, int direction,
48  int64_t start, int range, int curve);
49  void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
50  uint8_t * const *cf1,
51  int nb_samples, int channels,
52  int curve0, int curve1);
53 } AudioFadeContext;
54 
55 enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, NB_CURVES };
56 
57 #define OFFSET(x) offsetof(AudioFadeContext, x)
58 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
59 
60 static int query_formats(AVFilterContext *ctx)
61 {
62  AVFilterFormats *formats;
63  AVFilterChannelLayouts *layouts;
64  static const enum AVSampleFormat sample_fmts[] = {
65  AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
66  AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
67  AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
68  AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
69  AV_SAMPLE_FMT_NONE
70  };
71  int ret;
72 
73  layouts = ff_all_channel_counts();
74  if (!layouts)
75  return AVERROR(ENOMEM);
76  ret = ff_set_common_channel_layouts(ctx, layouts);
77  if (ret < 0)
78  return ret;
79 
80  formats = ff_make_format_list(sample_fmts);
81  if (!formats)
82  return AVERROR(ENOMEM);
83  ret = ff_set_common_formats(ctx, formats);
84  if (ret < 0)
85  return ret;
86 
87  formats = ff_all_samplerates();
88  if (!formats)
89  return AVERROR(ENOMEM);
90  return ff_set_common_samplerates(ctx, formats);
91 }
92 
93 static double fade_gain(int curve, int64_t index, int range)
94 {
95 #define CUBE(a) ((a)*(a)*(a))
96  double gain;
97 
98  gain = av_clipd(1.0 * index / range, 0, 1.0);
99 
100  switch (curve) {
101  case QSIN:
102  gain = sin(gain * M_PI / 2.0);
103  break;
104  case IQSIN:
105  /* 0.6... = 2 / M_PI */
106  gain = 0.6366197723675814 * asin(gain);
107  break;
108  case ESIN:
109  gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
110  break;
111  case HSIN:
112  gain = (1.0 - cos(gain * M_PI)) / 2.0;
113  break;
114  case IHSIN:
115  /* 0.3... = 1 / M_PI */
116  gain = 0.3183098861837907 * acos(1 - 2 * gain);
117  break;
118  case EXP:
119  /* -11.5... = 5*ln(0.1) */
120  gain = exp(-11.512925464970227 * (1 - gain));
121  break;
122  case LOG:
123  gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
124  break;
125  case PAR:
126  gain = 1 - sqrt(1 - gain);
127  break;
128  case IPAR:
129  gain = (1 - (1 - gain) * (1 - gain));
130  break;
131  case QUA:
132  gain *= gain;
133  break;
134  case CUB:
135  gain = CUBE(gain);
136  break;
137  case SQU:
138  gain = sqrt(gain);
139  break;
140  case CBR:
141  gain = cbrt(gain);
142  break;
143  case DESE:
144  gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
145  break;
146  case DESI:
147  gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
148  break;
149  }
150 
151  return gain;
152 }
153 
154 #define FADE_PLANAR(name, type) \
155 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
156  int nb_samples, int channels, int dir, \
157  int64_t start, int range, int curve) \
158 { \
159  int i, c; \
160  \
161  for (i = 0; i < nb_samples; i++) { \
162  double gain = fade_gain(curve, start + i * dir, range); \
163  for (c = 0; c < channels; c++) { \
164  type *d = (type *)dst[c]; \
165  const type *s = (type *)src[c]; \
166  \
167  d[i] = s[i] * gain; \
168  } \
169  } \
170 }
171 
172 #define FADE(name, type) \
173 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
174  int nb_samples, int channels, int dir, \
175  int64_t start, int range, int curve) \
176 { \
177  type *d = (type *)dst[0]; \
178  const type *s = (type *)src[0]; \
179  int i, c, k = 0; \
180  \
181  for (i = 0; i < nb_samples; i++) { \
182  double gain = fade_gain(curve, start + i * dir, range); \
183  for (c = 0; c < channels; c++, k++) \
184  d[k] = s[k] * gain; \
185  } \
186 }
187 
188 FADE_PLANAR(dbl, double)
189 FADE_PLANAR(flt, float)
190 FADE_PLANAR(s16, int16_t)
191 FADE_PLANAR(s32, int32_t)
192 
193 FADE(dbl, double)
194 FADE(flt, float)
195 FADE(s16, int16_t)
196 FADE(s32, int32_t)
197 
198 static int config_output(AVFilterLink *outlink)
199 {
200  AVFilterContext *ctx = outlink->src;
201  AudioFadeContext *s = ctx->priv;
202 
203  switch (outlink->format) {
204  case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
205  case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
206  case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
207  case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
208  case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
209  case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
210  case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
211  case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
212  }
213 
214  if (s->duration)
215  s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
216  if (s->start_time)
217  s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
218 
219  return 0;
220 }
221 
222 #if CONFIG_AFADE_FILTER
223 
224 static const AVOption afade_options[] = {
225  { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
226  { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
227  { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
228  { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
229  { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
230  { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
231  { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
232  { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
233  { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
234  { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
235  { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
236  { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
237  { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
238  { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
239  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
240  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
241  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
242  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
243  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
244  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
245  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
246  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
247  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
248  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
249  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
250  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
251  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
252  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
253  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
254  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
255  { NULL }
256 };
257 
258 AVFILTER_DEFINE_CLASS(afade);
259 
260 static av_cold int init(AVFilterContext *ctx)
261 {
262  AudioFadeContext *s = ctx->priv;
263 
264  if (INT64_MAX - s->nb_samples < s->start_sample)
265  return AVERROR(EINVAL);
266 
267  return 0;
268 }
269 
270 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
271 {
272  AudioFadeContext *s = inlink->dst->priv;
273  AVFilterLink *outlink = inlink->dst->outputs[0];
274  int nb_samples = buf->nb_samples;
275  AVFrame *out_buf;
276  int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
277 
278  if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
279  ( s->type && (cur_sample + nb_samples < s->start_sample)))
280  return ff_filter_frame(outlink, buf);
281 
282  if (av_frame_is_writable(buf)) {
283  out_buf = buf;
284  } else {
285  out_buf = ff_get_audio_buffer(inlink, nb_samples);
286  if (!out_buf)
287  return AVERROR(ENOMEM);
288  av_frame_copy_props(out_buf, buf);
289  }
290 
291  if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
292  ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
293  av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
294  av_frame_get_channels(out_buf), out_buf->format);
295  } else {
296  int64_t start;
297 
298  if (!s->type)
299  start = cur_sample - s->start_sample;
300  else
301  start = s->start_sample + s->nb_samples - cur_sample;
302 
303  s->fade_samples(out_buf->extended_data, buf->extended_data,
304  nb_samples, av_frame_get_channels(buf),
305  s->type ? -1 : 1, start,
306  s->nb_samples, s->curve);
307  }
308 
309  if (buf != out_buf)
310  av_frame_free(&buf);
311 
312  return ff_filter_frame(outlink, out_buf);
313 }
314 
315 static const AVFilterPad avfilter_af_afade_inputs[] = {
316  {
317  .name = "default",
318  .type = AVMEDIA_TYPE_AUDIO,
319  .filter_frame = filter_frame,
320  },
321  { NULL }
322 };
323 
324 static const AVFilterPad avfilter_af_afade_outputs[] = {
325  {
326  .name = "default",
327  .type = AVMEDIA_TYPE_AUDIO,
328  .config_props = config_output,
329  },
330  { NULL }
331 };
332 
333 AVFilter ff_af_afade = {
334  .name = "afade",
335  .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
336  .query_formats = query_formats,
337  .priv_size = sizeof(AudioFadeContext),
338  .init = init,
339  .inputs = avfilter_af_afade_inputs,
340  .outputs = avfilter_af_afade_outputs,
341  .priv_class = &afade_class,
342  .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
343 };
344 
345 #endif /* CONFIG_AFADE_FILTER */
346 
347 #if CONFIG_ACROSSFADE_FILTER
348 
349 static const AVOption acrossfade_options[] = {
350  { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
351  { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
352  { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS },
353  { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS },
354  { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
355  { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
356  { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
357  { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
358  { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
359  { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
360  { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
361  { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
362  { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
363  { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
364  { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
365  { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
366  { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
367  { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
368  { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
369  { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
370  { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
371  { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
372  { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
373  { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
374  { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
375  { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
376  { NULL }
377 };
378 
379 AVFILTER_DEFINE_CLASS(acrossfade);
380 
381 #define CROSSFADE_PLANAR(name, type) \
382 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
383  uint8_t * const *cf1, \
384  int nb_samples, int channels, \
385  int curve0, int curve1) \
386 { \
387  int i, c; \
388  \
389  for (i = 0; i < nb_samples; i++) { \
390  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
391  double gain1 = fade_gain(curve1, i, nb_samples); \
392  for (c = 0; c < channels; c++) { \
393  type *d = (type *)dst[c]; \
394  const type *s0 = (type *)cf0[c]; \
395  const type *s1 = (type *)cf1[c]; \
396  \
397  d[i] = s0[i] * gain0 + s1[i] * gain1; \
398  } \
399  } \
400 }
401 
402 #define CROSSFADE(name, type) \
403 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
404  uint8_t * const *cf1, \
405  int nb_samples, int channels, \
406  int curve0, int curve1) \
407 { \
408  type *d = (type *)dst[0]; \
409  const type *s0 = (type *)cf0[0]; \
410  const type *s1 = (type *)cf1[0]; \
411  int i, c, k = 0; \
412  \
413  for (i = 0; i < nb_samples; i++) { \
414  double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
415  double gain1 = fade_gain(curve1, i, nb_samples); \
416  for (c = 0; c < channels; c++, k++) \
417  d[k] = s0[k] * gain0 + s1[k] * gain1; \
418  } \
419 }
420 
421 CROSSFADE_PLANAR(dbl, double)
422 CROSSFADE_PLANAR(flt, float)
423 CROSSFADE_PLANAR(s16, int16_t)
424 CROSSFADE_PLANAR(s32, int32_t)
425 
426 CROSSFADE(dbl, double)
427 CROSSFADE(flt, float)
428 CROSSFADE(s16, int16_t)
429 CROSSFADE(s32, int32_t)
430 
431 static int acrossfade_filter_frame(AVFilterLink *inlink, AVFrame *in)
432 {
433  AVFilterContext *ctx = inlink->dst;
434  AudioFadeContext *s = ctx->priv;
435  AVFilterLink *outlink = ctx->outputs[0];
436  AVFrame *out, *cf[2] = { NULL };
437  int ret = 0, nb_samples;
438 
439  if (s->crossfade_is_over) {
440  in->pts = s->pts;
441  s->pts += av_rescale_q(in->nb_samples,
442  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
443  return ff_filter_frame(outlink, in);
444  } else if (inlink == ctx->inputs[0]) {
445  av_audio_fifo_write(s->fifo[0], (void **)in->extended_data, in->nb_samples);
446 
447  nb_samples = av_audio_fifo_size(s->fifo[0]) - s->nb_samples;
448  if (nb_samples > 0) {
449  out = ff_get_audio_buffer(outlink, nb_samples);
450  if (!out) {
451  ret = AVERROR(ENOMEM);
452  goto fail;
453  }
454  av_audio_fifo_read(s->fifo[0], (void **)out->extended_data, nb_samples);
455  out->pts = s->pts;
456  s->pts += av_rescale_q(nb_samples,
457  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
458  ret = ff_filter_frame(outlink, out);
459  }
460  } else if (av_audio_fifo_size(s->fifo[1]) < s->nb_samples) {
461  if (!s->overlap && av_audio_fifo_size(s->fifo[0]) > 0) {
462  nb_samples = av_audio_fifo_size(s->fifo[0]);
463 
464  cf[0] = ff_get_audio_buffer(outlink, nb_samples);
465  out = ff_get_audio_buffer(outlink, nb_samples);
466  if (!out || !cf[0]) {
467  ret = AVERROR(ENOMEM);
468  goto fail;
469  }
470  av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, nb_samples);
471 
472  s->fade_samples(out->extended_data, cf[0]->extended_data, nb_samples,
473  outlink->channels, -1, nb_samples - 1, nb_samples, s->curve);
474  out->pts = s->pts;
475  s->pts += av_rescale_q(nb_samples,
476  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
477  ret = ff_filter_frame(outlink, out);
478  if (ret < 0)
479  goto fail;
480  }
481 
482  av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
483  } else if (av_audio_fifo_size(s->fifo[1]) >= s->nb_samples) {
484  av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
485 
486  if (s->overlap) {
487  cf[0] = ff_get_audio_buffer(outlink, s->nb_samples);
488  cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
489  out = ff_get_audio_buffer(outlink, s->nb_samples);
490  if (!out || !cf[0] || !cf[1]) {
491  av_frame_free(&out);
492  ret = AVERROR(ENOMEM);
493  goto fail;
494  }
495 
496  av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, s->nb_samples);
497  av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
498 
499  s->crossfade_samples(out->extended_data, cf[0]->extended_data,
500  cf[1]->extended_data,
501  s->nb_samples, av_frame_get_channels(in),
502  s->curve, s->curve2);
503  out->pts = s->pts;
504  s->pts += av_rescale_q(s->nb_samples,
505  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
506  ret = ff_filter_frame(outlink, out);
507  if (ret < 0)
508  goto fail;
509  } else {
510  out = ff_get_audio_buffer(outlink, s->nb_samples);
511  cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
512  if (!out || !cf[1]) {
513  ret = AVERROR(ENOMEM);
514  av_frame_free(&out);
515  goto fail;
516  }
517 
518  av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
519 
520  s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
521  outlink->channels, 1, 0, s->nb_samples, s->curve2);
522  out->pts = s->pts;
523  s->pts += av_rescale_q(s->nb_samples,
524  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
525  ret = ff_filter_frame(outlink, out);
526  if (ret < 0)
527  goto fail;
528  }
529 
530  nb_samples = av_audio_fifo_size(s->fifo[1]);
531  if (nb_samples > 0) {
532  out = ff_get_audio_buffer(outlink, nb_samples);
533  if (!out) {
534  ret = AVERROR(ENOMEM);
535  goto fail;
536  }
537 
538  av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
539  out->pts = s->pts;
540  s->pts += av_rescale_q(nb_samples,
541  (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
542  ret = ff_filter_frame(outlink, out);
543  }
544  s->crossfade_is_over = 1;
545  }
546 
547 fail:
548  av_frame_free(&in);
549  av_frame_free(&cf[0]);
550  av_frame_free(&cf[1]);
551  return ret;
552 }
553 
554 static int acrossfade_request_frame(AVFilterLink *outlink)
555 {
556  AVFilterContext *ctx = outlink->src;
557  AudioFadeContext *s = ctx->priv;
558  int ret = 0;
559 
560  if (!s->cf0_eof) {
561  AVFilterLink *cf0 = ctx->inputs[0];
562  ret = ff_request_frame(cf0);
563  if (ret < 0 && ret != AVERROR_EOF)
564  return ret;
565  if (ret == AVERROR_EOF) {
566  s->cf0_eof = 1;
567  ret = 0;
568  }
569  } else {
570  AVFilterLink *cf1 = ctx->inputs[1];
571  int nb_samples = av_audio_fifo_size(s->fifo[1]);
572 
573  ret = ff_request_frame(cf1);
574  if (ret == AVERROR_EOF && nb_samples > 0) {
575  AVFrame *out = ff_get_audio_buffer(outlink, nb_samples);
576  if (!out)
577  return AVERROR(ENOMEM);
578 
579  av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
580  ret = ff_filter_frame(outlink, out);
581  }
582  }
583 
584  return ret;
585 }
586 
587 static int acrossfade_config_output(AVFilterLink *outlink)
588 {
589  AVFilterContext *ctx = outlink->src;
590  AudioFadeContext *s = ctx->priv;
591 
592  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
593  av_log(ctx, AV_LOG_ERROR,
594  "Inputs must have the same sample rate "
595  "%d for in0 vs %d for in1\n",
596  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
597  return AVERROR(EINVAL);
598  }
599 
600  outlink->sample_rate = ctx->inputs[0]->sample_rate;
601  outlink->time_base = ctx->inputs[0]->time_base;
602  outlink->channel_layout = ctx->inputs[0]->channel_layout;
603  outlink->channels = ctx->inputs[0]->channels;
604 
605  switch (outlink->format) {
606  case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
607  case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
608  case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
609  case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
610  case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
611  case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
612  case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
613  case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
614  }
615 
616  config_output(outlink);
617 
618  s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
619  s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
620  if (!s->fifo[0] || !s->fifo[1])
621  return AVERROR(ENOMEM);
622 
623  return 0;
624 }
625 
626 static av_cold void uninit(AVFilterContext *ctx)
627 {
628  AudioFadeContext *s = ctx->priv;
629 
630  av_audio_fifo_free(s->fifo[0]);
631  av_audio_fifo_free(s->fifo[1]);
632 }
633 
634 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
635  {
636  .name = "crossfade0",
637  .type = AVMEDIA_TYPE_AUDIO,
638  .filter_frame = acrossfade_filter_frame,
639  },
640  {
641  .name = "crossfade1",
642  .type = AVMEDIA_TYPE_AUDIO,
643  .filter_frame = acrossfade_filter_frame,
644  },
645  { NULL }
646 };
647 
648 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
649  {
650  .name = "default",
651  .type = AVMEDIA_TYPE_AUDIO,
652  .request_frame = acrossfade_request_frame,
653  .config_props = acrossfade_config_output,
654  },
655  { NULL }
656 };
657 
658 AVFilter ff_af_acrossfade = {
659  .name = "acrossfade",
660  .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
661  .query_formats = query_formats,
662  .priv_size = sizeof(AudioFadeContext),
663  .uninit = uninit,
664  .priv_class = &acrossfade_class,
665  .inputs = avfilter_af_acrossfade_inputs,
666  .outputs = avfilter_af_acrossfade_outputs,
667 };
668 
669 #endif /* CONFIG_ACROSSFADE_FILTER */
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
NULL
#define NULL
Definition: coverity.c:32
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:549
AV_OPT_TYPE_INT64
Definition: opt.h:224
s
const char * s
Definition: avisynth_c.h:768
CUBE
#define CUBE(a)
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
fade_gain
static double fade_gain(int curve, int64_t index, int range)
Definition: af_afade.c:93
av_audio_fifo_read
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:187
AVOption
AVOption.
Definition: opt.h:246
avfilter.h
Main libavfilter public API header.
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AV_OPT_TYPE_INT
Definition: opt.h:223
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_afade.c:198
AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
IHSIN
Definition: af_afade.c:55
src
#define src
Definition: vp8dsp.c:254
ESIN
Definition: af_afade.c:55
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:125
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilterContext::inputs
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:331
AV_OPT_TYPE_CONST
Definition: opt.h:232
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1125
start_time
static int64_t start_time
Definition: ffplay.c:328
uint8_t
uint8_t
Definition: audio_convert.c:194
av_cold
#define av_cold
Definition: attributes.h:82
uninit
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:287
opt.h
AVOptions.
AudioFadeContext::crossfade_samples
void(* crossfade_samples)(uint8_t **dst, uint8_t *const *cf0, uint8_t *const *cf1, int nb_samples, int channels, int curve0, int curve1)
Definition: af_afade.c:49
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_afade.c:60
FADE_PLANAR
#define FADE_PLANAR(name, type)
Definition: af_afade.c:154
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:271
duration
int64_t duration
Definition: movenc.c:63
AudioFadeContext::fade_samples
void(* fade_samples)(uint8_t **dst, uint8_t *const *src, int nb_samples, int channels, int direction, int64_t start, int range, int curve)
Definition: af_afade.c:46
flags
static int flags
Definition: log.c:57
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:62
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
ff_set_common_formats
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:568
av_samples_set_silence
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
CurveType
CurveType
Definition: af_afade.c:55
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
AVERROR
#define AVERROR(e)
Definition: error.h:43
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:163
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
AVFilterContext::priv
void * priv
private data for use by the filter
Definition: avfilter.h:338
cbrt
#define cbrt
Definition: tablegen.h:35
AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
fail
#define fail()
Definition: checkasm.h:89
exp
int8_t exp
Definition: eval.c:65
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
DESI
Definition: af_afade.c:55
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
av_rescale
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
AV_TIME_BASE
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:254
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
AudioFadeContext::start_sample
int64_t start_sample
Definition: af_afade.c:37
void
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
HSIN
Definition: af_afade.c:55
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: aeval.c:421
int32_t
int32_t
Definition: audio_convert.c:194
ctx
AVFormatContext * ctx
Definition: movenc.c:48
QUA
Definition: af_afade.c:55
AV_OPT_TYPE_BOOL
Definition: opt.h:240
AudioFadeContext::pts
int64_t pts
Definition: af_afade.c:44
AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
AudioFadeContext::crossfade_is_over
int crossfade_is_over
Definition: af_afade.c:42
outputs
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
AudioFadeContext::duration
int64_t duration
Definition: af_afade.c:38
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
NB_CURVES
Definition: af_afade.c:55
AVAudioFifo::nb_samples
int nb_samples
number of samples currently in the FIFO
Definition: audio_fifo.c:37
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:251
inputs
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:536
AudioFadeContext::fifo
AVAudioFifo * fifo[2]
Definition: af_afade.c:43
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
buf
void * buf
Definition: avisynth_c.h:690
CUB
Definition: af_afade.c:55
AudioFadeContext::start_time
int64_t start_time
Definition: af_afade.c:39
type
GLint GLenum type
Definition: opengl_enc.c:105
SQU
Definition: af_afade.c:55
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
av_frame_get_channels
int av_frame_get_channels(const AVFrame *frame)
AVFilter
Filter definition.
Definition: avfilter.h:144
index
int index
Definition: gxfenc.c:89
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
PAR
Definition: af_afade.c:55
FLAGS
#define FLAGS
Definition: af_afade.c:58
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
LOG
Definition: af_afade.c:55
DESE
Definition: af_afade.c:55
AVFilterContext::outputs
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:335
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
IPAR
Definition: af_afade.c:55
EXP
Definition: af_afade.c:55
AV_OPT_TYPE_DURATION
Definition: opt.h:237
TRI
Definition: af_afade.c:55
AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
OFFSET
#define OFFSET(x)
Definition: af_afade.c:57
QSIN
Definition: af_afade.c:55
audio_fifo.h
Audio FIFO Buffer.
AVFILTER_DEFINE_CLASS
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:342
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
AVFilterContext
An instance of a filter.
Definition: avfilter.h:323
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
out
FILE * out
Definition: movenc.c:54
start
void INT64 start
Definition: avisynth_c.h:690
AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
M_PI
#define M_PI
Definition: mathematics.h:52
ff_request_frame
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:405
formats
formats
Definition: signature.h:48
IQSIN
Definition: af_afade.c:55
internal.h
internal API functions
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:234
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:244
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:596
FADE
#define FADE(name, type)
Definition: af_afade.c:172
CBR
Definition: af_afade.c:55
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